[Asterisk-Users] Strange Sip (FWD, SipGate and such) problem

[Thomas Gallaway]

Karl Brose wrote:

> I think when you have this setup you need to keep the media path going 
> through Asterisk at all times.
> Your SIP is binding to both ports, internal and external, but that 
> doesn't correctly set it up for either scenario, localnet calls and 
> external calls. It won't keep the addresses straight for the RTP 
> channels.
> Try setting CANREINVITE=NO for peers (FWD,..) and for your local SIP 
> phones. When a channel is created in asterisk the media path is going 
> through Asterisk, but during a call the endpoints can issue reinvites 
> which switches the media path directly between the endpoints. You need 
> to prevent that.
> Other solutions are to run IAX to/from FWD and SIP locally, or SIP to 
> the external peers and IAX to a local IAX phone (or another protocol).
> Or you should be able to create your own NAT using the iptables and 
> bind asterisk only on one port either outside or inside and set the 
> right corresponding parameters. The RTP will still bind on all ports 
> currently, but that will be fixed in a matter of days.
>
> Also, sipgate.net should be sipgate.de (works ok though since they 
> don't care)
> fromdomain is meant to be realm not a hostname.
>
>
> Thomas Gallaway wrote:
>
>> Hi all
>>
>> I use sipgate and FWD but seem not to get it going. I do not have NAT 
>> on the asterisk box (static ip).
>> The asterisk box has 2 network interfaces. One internal and one 
>> external.
>>
>> Now when I make an call to a FWD or SipGate number all I get is
>>    -- Executing NoOp("SIP/113-6d2e", "") in new stack
>>    -- Executing Goto("SIP/113-6d2e", "intern-post|714551|1") in new 
>> stack
>>    -- Goto (intern-post,714551,1)
>>    -- Executing SetCallerID("SIP/113-6d2e", "270002") in new stack
>>    -- Executing SetCIDName("SIP/113-6d2e", "Thomas Gallaway") in new 
>> stack
>>    -- Executing Dial("SIP/113-6d2e", "SIP/14551 at fwd270002") in new stack
>>    -- Called 14551 at fwd270002
>>    -- SIP/fwd270002-6ee7 answered SIP/113-6d2e
>>    -- Attempting native bridge of SIP/113-6d2e and SIP/fwd270002-6ee7
>>  == Spawn extension (intern-post, 714551, 3) exited non-zero on 
>> 'SIP/113-6d2e'
>>
>> But either I get 1/2 second of audio or no audio. No matter how long 
>> I wait there is just no audio or just a short snippet of audio at the 
>> beginning.
>>
>> Here is parts of my sip.conf;
>> [general]
>> port = 5060                     ; Port to bind to
>> localnet = 192.168.1.0         ; Internal NETWORK address
>> localmask = 255.255.255.0      ; Internal netmask
>> externip = 206.40.161.235
>> context = intern                ; Default for incoming calls
>> maxexpirey=3600
>> defaultexpirey=300
>> disallow=all                    ; Disallow all codecsa
>> allow=gsm
>> allow=alaw
>> allow=ulaw
>> tos=reliability
>> register => xxx:xxx at sipgate.de/150
>> register => xxx:xxx at fwd.pulver.com/151
>>
>>
>> [sipgate1]
>> type=friend
>> username=xxx
>> secret=xxx
>> host=sipgate.de
>> fromuser=xxx
>> fromdomain=sipgate.net
>> nat=no
>> context=incoming-sipgate
>> canreinvite=yes
>>
>> [fwd270002]
>> allow=ulaw
>> type=friend
>> context=incoming-fwd
>> secret=xxx
>> username=xxx
>> host=fwd.pulver.com
>>
>> Any ideas?
>> When I put nat=yes I actually will get 1 second of audio, then it dies.
>> I have been googling for a while now and not seem to find any 
>> sollution to this.
>>
>> -- Thomas
>
I will try that. I had to remove all the IAX / SIP changes I did as even 
on the local network it started to give me a one way communication 
thing. I was able to hear other people but they could not hear me.

Will give this another try later in the week.

-- Thomas